commit af512281b1b35759155b4151e8bf83467c28533c
author Philipp Hancke <phancke@microsoft.com> Fri Oct 14 07:32:58 2022
committer WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> Tue Nov 01 11:46:49 2022
tree ef558e729f924fa2b740c3b2bd23e58336e18c9b
parent 1cb3cdece5e47b774e8ba5da9d78fa86b39f10e5

audio: make packets lost a signed integer

as it is defined in RFC 3550. This avoids implicit casts
between signed and unsigned definitions.

BUG=webrtc:8626

Change-Id: I919b7c38ede1aa8d32f8e31b55660f540e5f5a6b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279240
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#38522}

audio/channel_receive.h

diff --git a/audio/channel_receive.h b/audio/channel_receive.h
index c3eca29..b47a4b5 100644
--- a/audio/channel_receive.h
+++ b/audio/channel_receive.h
@@ -51,7 +51,7 @@
 class RtpRtcp;
 
 struct CallReceiveStatistics {
-  unsigned int cumulativeLost;
+  int cumulativeLost;
   unsigned int jitterSamples;
   int64_t payload_bytes_rcvd = 0;
   int64_t header_and_padding_bytes_rcvd = 0;

audio/test/audio_stats_test.cc

diff --git a/audio/test/audio_stats_test.cc b/audio/test/audio_stats_test.cc
index febcb06..c637bff 100644
--- a/audio/test/audio_stats_test.cc
+++ b/audio/test/audio_stats_test.cc
@@ -68,7 +68,7 @@
         receive_stream()->GetStats(/*get_and_clear_legacy_stats=*/true);
     EXPECT_PRED2(IsNear, kBytesSent, recv_stats.payload_bytes_rcvd);
     EXPECT_PRED2(IsNear, kPacketsSent, recv_stats.packets_rcvd);
-    EXPECT_EQ(0u, recv_stats.packets_lost);
+    EXPECT_EQ(0, recv_stats.packets_lost);
     EXPECT_EQ("opus", send_stats.codec_name);
     // recv_stats.jitter_ms
     // recv_stats.jitter_buffer_ms

call/audio_receive_stream.h

diff --git a/call/audio_receive_stream.h b/call/audio_receive_stream.h
index 5d3c38f..f383277 100644
--- a/call/audio_receive_stream.h
+++ b/call/audio_receive_stream.h
@@ -38,7 +38,7 @@
     uint32_t packets_rcvd = 0;
     uint64_t fec_packets_received = 0;
     uint64_t fec_packets_discarded = 0;
-    uint32_t packets_lost = 0;
+    int32_t packets_lost = 0;
     uint64_t packets_discarded = 0;
     uint32_t nacks_sent = 0;
     std::string codec_name;

media/engine/webrtc_voice_engine_unittest.cc

diff --git a/media/engine/webrtc_voice_engine_unittest.cc b/media/engine/webrtc_voice_engine_unittest.cc
index f61cfb9..9644fbd 100644
--- a/media/engine/webrtc_voice_engine_unittest.cc
+++ b/media/engine/webrtc_voice_engine_unittest.cc
@@ -707,8 +707,7 @@
               stats.header_and_padding_bytes_rcvd);
     EXPECT_EQ(rtc::checked_cast<unsigned int>(info.packets_rcvd),
               stats.packets_rcvd);
-    EXPECT_EQ(rtc::checked_cast<unsigned int>(info.packets_lost),
-              stats.packets_lost);
+    EXPECT_EQ(info.packets_lost, stats.packets_lost);
     EXPECT_EQ(info.codec_name, stats.codec_name);
     EXPECT_EQ(info.codec_payload_type, stats.codec_payload_type);
     EXPECT_EQ(rtc::checked_cast<unsigned int>(info.jitter_ms), stats.jitter_ms);